U.S. Pat. No. 7,457,757 B1 describes a method of increasing speech intelligibility of acoustic sounds recorded with a hearing aid, wherein an incoming signal is processed according to an adaptive algorithm. The incoming signal is fed to a signal processing stage that comprises a low pass filter, a high pass filter, an expander, a compressor and a pass band contour, wherein these components can be adaptively controlled, in particular turned on or off, through the adaption algorithm. It is described that a modulation depth of the incoming signal is determined by using an intelligibility measurement in order to obtain an estimation of a signal to noise ratio of the incoming signal. The adapted algorithm is used for computing and/or for choosing the best configuration parameters such that the incoming signal is optimally processed.
Even though the incoming signal is processed in dependence of a modulation depth of the input signal, speech intelligibility can be unsatisfactory, in particular for a hearing impaired person that can perceive sound pressure levels in a substantially decreased dynamic range only. WO 2006/133431 A2 describes a method of improving the naturalness of processed sound by separating the information-bearing spectral envelope from the voice-quality-bearing spectral fine structure. The spectral envelope (formants) are estimated in real time and shifted to a higher frequency range, whereas the fine structure is kept intact.